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	 <pre class="pre meta-info">[<a href="https://datatracker.ietf.org" title="Document search and retrieval page">Search</a>] [<a href="https://www.rfc-editor.org/rfc/rfc8108.txt" title="Plaintext version of this document">txt</a>|<a href="https://www.rfc-editor.org/rfc/rfc8108.html" title="HTML version of this document, from XML2RFC">html</a>|<a href="https://www.rfc-editor.org/rfc/pdfrfc/rfc8108.txt.pdf" title="PDF version of this document">pdf</a>|<a href="/doc/rfc8108/bibtex/" title="BibTex entry for this document">bibtex</a>] [<a href="/doc/rfc8108/" title="Datatracker information for this document">Tracker</a>] [<a href="/group/avtcore/" title="The working group handling this document">WG</a>] [<a href="mailto:draft-ietf-avtcore-rtp-multi-stream@ietf.org?subject=draft-ietf-avtcore-rtp-multi-stream" title="Send email to the document authors">Email</a>] [<a href="https://www.ietf.org/rfcdiff?difftype=--hwdiff&url2=draft-ietf-avtcore-rtp-multi-stream-11.txt" title="Inline diff (wdiff)">Diff1</a>] [<a href="https://www.ietf.org/rfcdiff?url2=draft-ietf-avtcore-rtp-multi-stream-11.txt" title="Side-by-side diff">Diff2</a>] [<a href="https://www.ietf.org/tools/idnits?url=https://www.ietf.org/archive/id/draft-ietf-avtcore-rtp-multi-stream-11.txt" title="Run an idnits check of this document">Nits</a>]

From: <a href="/doc/html/draft-ietf-avtcore-rtp-multi-stream-11">draft-ietf-avtcore-rtp-multi-stream-11</a>           Proposed Standard</pre>
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    <div class="draftcontent">
    <pre>Internet Engineering Task Force (IETF)                         J. Lennox
Request for Comments: 8108                                         Vidyo
Updates: <a href="/doc/html/rfc3550">3550</a>, <a href="/doc/html/rfc4585">4585</a>                                        M. Westerlund
Category: Standards Track                                       Ericsson
ISSN: 2070-1721                                                    Q. Wu
                                                                  Huawei
                                                              C. Perkins
                                                   University of Glasgow
                                                              March 2017


          <span class="h1">Sending Multiple RTP Streams in a Single RTP Session</span>

Abstract

   This memo expands and clarifies the behavior of Real-time Transport
   Protocol (RTP) endpoints that use multiple synchronization sources
   (SSRCs).  This occurs, for example, when an endpoint sends multiple
   RTP streams in a single RTP session.  This memo updates <a href="/doc/html/rfc3550">RFC 3550</a> with
   regard to handling multiple SSRCs per endpoint in RTP sessions, with
   a particular focus on RTP Control Protocol (RTCP) behavior.  It also
   updates <a href="/doc/html/rfc4585">RFC 4585</a> to change and clarify the calculation of the timeout
   of SSRCs and the inclusion of feedback messages.

Status of This Memo

   This is an Internet Standards Track document.

   This document is a product of the Internet Engineering Task Force
   (IETF).  It represents the consensus of the IETF community.  It has
   received public review and has been approved for publication by the
   Internet Engineering Steering Group (IESG).  Further information on
   Internet Standards is available in <a href="/doc/html/rfc7841#section-2">Section&nbsp;2 of RFC 7841</a>.

   Information about the current status of this document, any errata,
   and how to provide feedback on it may be obtained at
   <a href="https://www.rfc-editor.org/info/rfc8108">http://www.rfc-editor.org/info/rfc8108</a>.














<span class="grey">Lennox, et al.               Standards Track                    [Page 1]</span></pre>
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<span class="grey"><a href="/doc/html/rfc8108">RFC 8108</a>        Multiple Media Streams in an RTP Session      March 2017</span>


Copyright Notice

   Copyright (c) 2017 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to <a href="/doc/html/bcp78">BCP 78</a> and the IETF Trust&#x27;s Legal
   Provisions Relating to IETF Documents
   (<a href="https://trustee.ietf.org/license-info">http://trustee.ietf.org/license-info</a>) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.





































<span class="grey">Lennox, et al.               Standards Track                    [Page 2]</span></pre>
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<span class="grey"><a href="/doc/html/rfc8108">RFC 8108</a>        Multiple Media Streams in an RTP Session      March 2017</span>


Table of Contents

   <a href="#section-1">1</a>. Introduction ....................................................<a href="#page-4">4</a>
   <a href="#section-2">2</a>. Terminology .....................................................<a href="#page-4">4</a>
   <a href="#section-3">3</a>. Use Cases for Multi-Stream Endpoints ............................<a href="#page-4">4</a>
      <a href="#section-3.1">3.1</a>. Endpoints with Multiple Capture Devices ....................<a href="#page-4">4</a>
      <a href="#section-3.2">3.2</a>. Multiple Media Types in a Single RTP Session ...............<a href="#page-5">5</a>
      <a href="#section-3.3">3.3</a>. Multiple Stream Mixers .....................................<a href="#page-5">5</a>
      <a href="#section-3.4">3.4</a>. Multiple SSRCs for a Single Media Source ...................<a href="#page-5">5</a>
   <a href="#section-4">4</a>. Use of RTP by Endpoints That Send Multiple Media Streams ........<a href="#page-6">6</a>
   <a href="#section-5">5</a>. Use of RTCP by Endpoints That Send Multiple Media Streams .......<a href="#page-6">6</a>
      <a href="#section-5.1">5.1</a>. RTCP Reporting Requirement .................................<a href="#page-7">7</a>
      <a href="#section-5.2">5.2</a>. Initial Reporting Interval .................................<a href="#page-7">7</a>
      <a href="#section-5.3">5.3</a>. Aggregation of Reports into Compound RTCP Packets ..........<a href="#page-8">8</a>
           <a href="#section-5.3.1">5.3.1</a>. Maintaining AVG_RTCP_SIZE ...........................<a href="#page-9">9</a>
           <a href="#section-5.3.2">5.3.2</a>. Scheduling RTCP when Aggregating Multiple SSRCs ....<a href="#page-10">10</a>
      <a href="#section-5.4">5.4</a>. Use of RTP/AVPF or RTP/SAVPF Feedback .....................<a href="#page-13">13</a>
           <a href="#section-5.4.1">5.4.1</a>. Choice of SSRC for Feedback Packets ................<a href="#page-13">13</a>
           <a href="#section-5.4.2">5.4.2</a>. Scheduling an RTCP Feedback Packet .................<a href="#page-14">14</a>
   <a href="#section-6">6</a>. Adding and Removing SSRCs ......................................<a href="#page-15">15</a>
      <a href="#section-6.1">6.1</a>. Adding RTP Streams ........................................<a href="#page-16">16</a>
      <a href="#section-6.2">6.2</a>. Removing RTP Streams ......................................<a href="#page-16">16</a>
   <a href="#section-7">7</a>. RTCP Considerations for Streams with Disparate Rates ...........<a href="#page-17">17</a>
      <a href="#section-7.1">7.1</a>. Timing Out SSRCs ..........................................<a href="#page-19">19</a>
           7.1.1. Problems with the RTP/AVPF T_rr_interval
                  Parameter ..........................................<a href="#page-19">19</a>
           <a href="#section-7.1.2">7.1.2</a>. Avoiding Premature Timeout .........................<a href="#page-20">20</a>
           <a href="#section-7.1.3">7.1.3</a>. Interoperability between RTP/AVP and RTP/AVPF ......<a href="#page-21">21</a>
           <a href="#section-7.1.4">7.1.4</a>. Updated SSRC Timeout Rules .........................<a href="#page-22">22</a>
      <a href="#section-7.2">7.2</a>. Tuning RTCP Transmissions .................................<a href="#page-22">22</a>
           <a href="#section-7.2.1">7.2.1</a>. RTP/AVP and RTP/SAVP ...............................<a href="#page-22">22</a>
           <a href="#section-7.2.2">7.2.2</a>. RTP/AVPF and RTP/SAVPF .............................<a href="#page-24">24</a>
   <a href="#section-8">8</a>. Security Considerations ........................................<a href="#page-25">25</a>
   <a href="#section-9">9</a>. References .....................................................<a href="#page-26">26</a>
      <a href="#section-9.1">9.1</a>. Normative References ......................................<a href="#page-26">26</a>
      <a href="#section-9.2">9.2</a>. Informative References ....................................<a href="#page-26">26</a>
   Acknowledgments ...................................................<a href="#page-29">29</a>
   Authors&#x27; Addresses ................................................<a href="#page-29">29</a>













<span class="grey">Lennox, et al.               Standards Track                    [Page 3]</span></pre>
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<span class="grey"><a href="/doc/html/rfc8108">RFC 8108</a>        Multiple Media Streams in an RTP Session      March 2017</span>


<span class="h2"><a class="selflink" id="section-1" href="#section-1">1</a>.  Introduction</span>

   At the time the Real-Time Transport Protocol (RTP) [<a href="/doc/html/rfc3550" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">RFC3550</a>] was
   originally designed, and for quite some time after, endpoints in RTP
   sessions typically only transmitted a single media source and, thus,
   used a single RTP stream and synchronization source (SSRC) per RTP
   session, where separate RTP sessions were typically used for each
   distinct media type.  Recently, however, a number of scenarios have
   emerged in which endpoints wish to send multiple RTP streams,
   distinguished by distinct RTP synchronization source (SSRC)
   identifiers, in a single RTP session.  These are outlined in
   <a href="#section-3">Section 3</a>.  Although the initial design of RTP did consider such
   scenarios, the specification was not consistently written with such
   use cases in mind; thus, the specification is somewhat unclear in
   places.

   This memo updates [<a href="/doc/html/rfc3550" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">RFC3550</a>] to clarify behavior in use cases where
   endpoints use multiple SSRCs.  It also updates [<a href="/doc/html/rfc4585" title="&quot;Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)&quot;">RFC4585</a>] to resolve
   problems with regard to timeout of inactive SSRCs and to clarify
   behavior around inclusion of feedback messages.

<span class="h2"><a class="selflink" id="section-2" href="#section-2">2</a>.  Terminology</span>

   The key words &quot;MUST&quot;, &quot;MUST NOT&quot;, &quot;REQUIRED&quot;, &quot;SHALL&quot;, &quot;SHALL NOT&quot;,
   &quot;SHOULD&quot;, &quot;SHOULD NOT&quot;, &quot;RECOMMENDED&quot;, &quot;NOT RECOMMENDED&quot;, &quot;MAY&quot;, and
   &quot;OPTIONAL&quot; in this document are to be interpreted as described in <a href="/doc/html/rfc2119">RFC</a>
   <a href="/doc/html/rfc2119">2119</a> [<a href="/doc/html/rfc2119" title="&quot;Key words for use in RFCs to Indicate Requirement Levels&quot;">RFC2119</a>] and indicate requirement levels for compliant
   implementations.

<span class="h2"><a class="selflink" id="section-3" href="#section-3">3</a>.  Use Cases for Multi-Stream Endpoints</span>

   This section discusses several use cases that have motivated the
   development of endpoints that sends RTP data using multiple SSRCs in
   a single RTP session.

<span class="h3"><a class="selflink" id="section-3.1" href="#section-3.1">3.1</a>.  Endpoints with Multiple Capture Devices</span>

   The most straightforward motivation for an endpoint to send multiple
   simultaneous RTP streams in a single RTP session is when an endpoint
   has multiple capture devices and, hence, can generate multiple media
   sources, of the same media type and characteristics.  For example,
   telepresence systems of the type described by the CLUE Telepresence
   Framework [<a href="#ref-CLUE-FRAME">CLUE-FRAME</a>] often have multiple cameras or microphones
   covering various areas of a room and, hence, send several RTP streams
   of each type within a single RTP session.






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<span class="grey"><a href="/doc/html/rfc8108">RFC 8108</a>        Multiple Media Streams in an RTP Session      March 2017</span>


<span class="h3"><a class="selflink" id="section-3.2" href="#section-3.2">3.2</a>.  Multiple Media Types in a Single RTP Session</span>

   Recent work has updated RTP [<a href="#ref-MULTI-RTP">MULTI-RTP</a>] and Session Description
   Protocol (SDP) [<a href="#ref-SDP-BUNDLE">SDP-BUNDLE</a>] to remove the historical assumption in
   RTP that media sources of different media types would always be sent
   on different RTP sessions.  In this work, a single endpoint&#x27;s audio
   and video RTP streams (for example) are instead sent in a single RTP
   session to reduce the number of transport-layer flows used.

<span class="h3"><a class="selflink" id="section-3.3" href="#section-3.3">3.3</a>.  Multiple Stream Mixers</span>

   There are several RTP topologies that can involve a central device
   that itself generates multiple RTP streams in a session.  An example
   is a mixer providing centralized compositing for a multi-capture
   scenario like that described in <a href="#section-3.1">Section 3.1</a>.  In this case, the
   centralized node is behaving much like a multi-capturer endpoint,
   generating several similar and related sources.

   A more complex example is the selective forwarding middlebox,
   described in <a href="/doc/html/rfc7667#section-3.7">Section&nbsp;3.7 of [RFC7667]</a>.  This is a middlebox that
   receives RTP streams from several endpoints and then selectively
   forwards modified versions of some RTP streams toward the other
   endpoints to which it is connected.  For each connected endpoint, a
   separate media source appears in the session for every other source
   connected to the middlebox, &quot;projected&quot; from the original streams,
   but at any given time many of them can appear to be inactive (and
   thus are receivers, not senders, in RTP).  This sort of device is
   closer to being an RTP mixer than an RTP translator: it terminates
   RTCP reporting about the mixed streams; it can rewrite SSRCs,
   timestamps, and sequence numbers, as well as the contents of the RTP
   payloads; and it can turn sources on and off at will without
   appearing to generate packet loss.  Each projected stream will
   typically preserve its original RTCP source description (SDES)
   information.

<span class="h3"><a class="selflink" id="section-3.4" href="#section-3.4">3.4</a>.  Multiple SSRCs for a Single Media Source</span>

   There are also several cases where multiple SSRCs can be used to send
   data from a single media source within a single RTP session.  These
   include, but are not limited to, transport robustness tools, such as
   the RTP retransmission payload format [<a href="/doc/html/rfc4588" title="&quot;RTP Retransmission Payload Format&quot;">RFC4588</a>], that require one
   SSRC to be used for the media data and another SSRC for the repair
   data.  Similarly, some layered media encoding schemes, for example,
   H.264 Scalable Video Coding (SVC) [<a href="/doc/html/rfc6190" title="&quot;RTP Payload Format for Scalable Video Coding&quot;">RFC6190</a>], can be used in a
   configuration where each layer is sent using a different SSRC within
   a single RTP session.





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<span class="grey"><a href="/doc/html/rfc8108">RFC 8108</a>        Multiple Media Streams in an RTP Session      March 2017</span>


<span class="h2"><a class="selflink" id="section-4" href="#section-4">4</a>.  Use of RTP by Endpoints That Send Multiple Media Streams</span>

   RTP is inherently a group communication protocol.  Each endpoint in
   an RTP session will use one or more SSRCs, as will some types of RTP-
   level middlebox.  Accordingly, unless restrictions on the number of
   SSRCs have been signaled, RTP endpoints can expect to receive RTP
   data packets sent using a number of different SSRCs, within a single
   RTP session.  This can occur irrespective of whether the RTP session
   is running over a point-to-point connection or a multicast group,
   since middleboxes can be used to connect multiple transport
   connections together into a single RTP session (the RTP session is
   defined by the shared SSRC space, not by the transport connections).
   Furthermore, if RTP mixers are used, some SSRCs might only be visible
   in the contributing source (CSRC) list of an RTP packet and in RTCP,
   and might not appear directly as the SSRC of an RTP data packet.

   Every RTP endpoint will have an allocated share of the available
   session bandwidth, as determined by signaling and congestion control.
   The endpoint needs to keep its total media sending rate within this
   share.  However, endpoints that send multiple RTP streams do not
   necessarily need to subdivide their share of the available bandwidth
   independently or uniformly to each RTP stream and its SSRCs.  In
   particular, an endpoint can vary the bandwidth allocation to
   different streams depending on their needs, and it can dynamically
   change the bandwidth allocated to different SSRCs (for example, by
   using a variable-rate codec), provided the total sending rate does
   not exceed its allocated share.  This includes enabling or disabling
   RTP streams, or their redundancy streams, as more or less bandwidth
   becomes available.

<span class="h2"><a class="selflink" id="section-5" href="#section-5">5</a>.  Use of RTCP by Endpoints That Send Multiple Media Streams</span>

   RTCP is defined in <a href="/doc/html/rfc3550#section-6">Section&nbsp;6 of [RFC3550]</a>.  The description of the
   protocol is phrased in terms of the behavior of &quot;participants&quot; in an
   RTP session, under the assumption that each endpoint is a participant
   with a single SSRC.  However, for correct operation in cases where
   endpoints have multiple SSRC values, implementations MUST treat each
   SSRC as a separate participant in the RTP session, so that an
   endpoint that has multiple SSRCs counts as multiple participants.












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<span class="grey"><a href="/doc/html/rfc8108">RFC 8108</a>        Multiple Media Streams in an RTP Session      March 2017</span>


<span class="h3"><a class="selflink" id="section-5.1" href="#section-5.1">5.1</a>.  RTCP Reporting Requirement</span>

   An RTP endpoint that has multiple SSRCs MUST treat each SSRC as a
   separate participant in the RTP session.  Each SSRC will maintain its
   own RTCP-related state information and, hence, will have its own RTCP
   reporting interval that determines when it sends RTCP reports.  If
   the mechanism in [<a href="#ref-MULTI-STREAM-OPT">MULTI-STREAM-OPT</a>] is not used, then each SSRC will
   send RTCP reports for all other SSRCs, including those co-located at
   the same endpoint.

   If the endpoint has some SSRCs that are sending data and some that
   are only receivers, then they will receive different shares of the
   RTCP bandwidth and calculate different base RTCP reporting intervals.
   Otherwise, all SSRCs at an endpoint will calculate the same base RTCP
   reporting interval.  The actual reporting intervals for each SSRC are
   randomized in the usual way, but reports can be aggregated as
   described in <a href="#section-5.3">Section 5.3</a>.

<span class="h3"><a class="selflink" id="section-5.2" href="#section-5.2">5.2</a>.  Initial Reporting Interval</span>

   When a participant joins a unicast session, the following text from
   <a href="/doc/html/rfc3550#section-6.2">Section&nbsp;6.2 of [RFC3550]</a> is relevant: &quot;For unicast sessions... the
   delay before sending the initial compound RTCP packet MAY be zero.&quot;
   The basic assumption is that this also ought to apply in the case of
   multiple SSRCs.  Caution has to be exercised, however, when an
   endpoint (or middlebox) with a large number of SSRCs joins a unicast
   session, since immediate transmission of many RTCP reports can create
   a significant burst of traffic, leading to transient congestion and
   packet loss due to queue overflows.

   To ensure that the initial burst of traffic generated by an RTP
   endpoint is no larger than would be generated by a TCP connection, an
   RTP endpoint MUST NOT send more than four compound RTCP packets with
   zero initial delay when it joins an RTP session, independent of the
   number of SSRCs used by the endpoint.  Each of those initial compound
   RTCP packets MAY include aggregated reports from multiple SSRCs,
   provided the total compound RTCP packet size does not exceed the MTU,
   and the avg_rtcp_size is maintained as in <a href="#section-5.3.1">Section 5.3.1</a>.  Aggregating
   reports from several SSRCs in the initial compound RTCP packets
   allows a substantial number of SSRCs to report immediately.
   Endpoints SHOULD prioritize reports on SSRCs that are likely to be
   most immediately useful, e.g., for SSRCs that are initially senders.

   An endpoint that needs to report on more SSRCs than will fit into the
   four compound RTCP reports that can be sent immediately MUST send the
   other reports later, following the usual RTCP timing rules including
   timer reconsideration.  Those reports MAY be aggregated as described
   in <a href="#section-5.3">Section 5.3</a>.



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      Note: The above is chosen to match the TCP maximum initial window
      of four packets [<a href="/doc/html/rfc3390" title="&quot;Increasing TCP&#x27;s Initial Window&quot;">RFC3390</a>], not the larger TCP initial windows for
      which there is an ongoing experiment [<a href="/doc/html/rfc6928" title="&quot;Increasing TCP&#x27;s Initial Window&quot;">RFC6928</a>].  The reason for
      this is a desire to be conservative, since an RTP endpoint will
      also in many cases start sending RTP data packets at the same time
      as these initial RTCP packets are sent.

<span class="h3"><a class="selflink" id="section-5.3" href="#section-5.3">5.3</a>.  Aggregation of Reports into Compound RTCP Packets</span>

   As outlined in <a href="#section-5.1">Section 5.1</a>, an endpoint with multiple SSRCs has to
   treat each SSRC as a separate participant when it comes to sending
   RTCP reports.  This will lead to each SSRC sending a compound RTCP
   packet in each reporting interval.  Since these packets are coming
   from the same endpoint, it might reasonably be expected that they can
   be aggregated to reduce overheads.  Indeed, <a href="/doc/html/rfc3550#section-6.1">Section&nbsp;6.1 of [RFC3550]</a>
   allows RTP translators and mixers to aggregate packets in similar
   circumstances:

      It is RECOMMENDED that translators and mixers combine individual
      RTCP packets from the multiple sources they are forwarding into
      one compound packet whenever feasible in order to amortize the
      packet overhead (see <a href="#section-7">Section 7</a>).  An example RTCP compound packet
      as might be produced by a mixer is shown in Fig. 1.  If the
      overall length of a compound packet would exceed the MTU of the
      network path, it SHOULD be segmented into multiple shorter
      compound packets to be transmitted in separate packets of the
      underlying protocol.  This does not impair the RTCP bandwidth
      estimation because each compound packet represents at least one
      distinct participant.  Note that each of the compound packets MUST
      begin with an SR or RR packet.

   This allows RTP translators and mixers to generate compound RTCP
   packets that contain multiple Sender Report (SR) or Receiver Report
   (RR) packets from different SSRCs, as well as any of the other packet
   types.  There are no restrictions on the order in which the RTCP
   packets can occur within the compound packet, except the regular rule
   that the compound RTCP packet starts with an SR or RR packet.  Due to
   this rule, correctly implemented RTP endpoints will be able to handle
   compound RTCP packets that contain RTCP packets relating to multiple
   SSRCs.

   Accordingly, endpoints that use multiple SSRCs can aggregate the RTCP
   packets sent by their different SSRCs into compound RTCP packets,
   provided 1) the resulting compound RTCP packets begin with an SR or
   RR packet, 2) they maintain the average RTCP packet size as described
   in <a href="#section-5.3.1">Section 5.3.1</a>, and 3) they schedule packet transmission and manage
   aggregation as described in <a href="#section-5.3.2">Section 5.3.2</a>.




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<span class="h4"><a class="selflink" id="section-5.3.1" href="#section-5.3.1">5.3.1</a>.  Maintaining AVG_RTCP_SIZE</span>

   The RTCP scheduling algorithm in [<a href="/doc/html/rfc3550" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">RFC3550</a>] works on a per-SSRC basis.
   Each SSRC sends a single compound RTCP packet in each RTCP reporting
   interval.  When an endpoint uses multiple SSRCs, it is desirable to
   aggregate the compound RTCP packets sent by its SSRCs, reducing the
   overhead by forming a larger compound RTCP packet.  This aggregation
   can be done as described in <a href="#section-5.3.2">Section 5.3.2</a>, provided the average RTCP
   packet size calculation is updated as follows.

   Participants in an RTP session update their estimate of the average
   RTCP packet size (avg_rtcp_size) each time they send or receive an
   RTCP packet (see <a href="/doc/html/rfc3550#section-6.3.3">Section&nbsp;6.3.3 of [RFC3550]</a>).  When a compound RTCP
   packet that contains RTCP packets from several SSRCs is sent or
   received, the avg_rtcp_size estimate for each SSRC that is reported
   upon is updated using div_packet_size rather than the actual packet
   size:

      avg_rtcp_size = (1/16) * div_packet_size + (15/16) * avg_rtcp_size

   where div_packet_size is packet_size divided by the number of SSRCs
   reporting in that compound packet.  The number of SSRCs reporting in
   a compound packet is determined by counting the number of different
   SSRCs that are the source of SR or RR RTCP packets within the
   compound RTCP packet.  Non-compound RTCP packets (i.e., RTCP packets
   that do not contain an SR or RR packet [<a href="/doc/html/rfc5506" title="&quot;Support for Reduced-Size Real-Time Transport Control Protocol (RTCP): Opportunities and Consequences&quot;">RFC5506</a>]) are considered to
   report on a single SSRC.

   A participant that doesn&#x27;t follow the above rule, and instead uses
   the full RTCP compound packet size to calculate avg_rtcp_size, will
   derive an RTCP reporting interval that is overly large by a factor
   that is proportional to the number of SSRCs aggregated into compound
   RTCP packets and the size of set of SSRCs being aggregated relative
   to the total number of participants.  This increased RTCP reporting
   interval can cause premature timeouts if it is more than five times
   the interval chosen by the SSRCs that understand compound RTCP that
   aggregate reports from many SSRCs.  A 1500-octet MTU can fit five
   typical-size reports into a compound RTCP packet, so this is a real
   concern if endpoints aggregate RTCP reports from multiple SSRCs.

   The issue raised in the previous paragraph is mitigated by the
   modification in timeout behavior specified in <a href="#section-7.1.2">Section 7.1.2</a> of this
   memo.  This mitigation is in place in those cases where the RTCP
   bandwidth is sufficiently high that an endpoint, using avg_rtcp_size
   calculated without taking into account the number of reporting SSRCs,
   can transmit more frequently than approximately every 5 seconds.
   Note, however, that the non-updated endpoint&#x27;s RTCP reporting is
   still negatively impacted even if the premature timeouts of its SSRCs



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   are avoided.  If compatibility with non-updated endpoints is a
   concern, the number of reports from different SSRCs aggregated into a
   single compound RTCP packet SHOULD either be limited to two reports
   or aggregation ought not be used at all.  This will limit the
   non-updated endpoint&#x27;s RTCP reporting interval to be no larger than
   twice the RTCP reporting interval that would be chosen by an endpoint
   following this specification.

<span class="h4"><a class="selflink" id="section-5.3.2" href="#section-5.3.2">5.3.2</a>.  Scheduling RTCP when Aggregating Multiple SSRCs</span>

   This section revises and extends the behavior defined in <a href="/doc/html/rfc3550#section-6.3">Section&nbsp;6.3
   of [RFC3550]</a>, and in <a href="/doc/html/rfc4585#section-3.5.3">Section&nbsp;3.5.3 of [RFC4585]</a> if the RTP/AVPF
   profile or the RTP/SAVPF profile is used, regarding actions to take
   when scheduling and sending RTCP packets where multiple reporting
   SSRCs are aggregating their RTCP packets into the same compound RTCP
   packet.  These changes to the RTCP scheduling rules are needed to
   maintain important RTCP timing properties, including the inter-packet
   distribution, and the behavior during flash joins and other changes
   in session membership.

   The variables tn, tp, tc, T, and Td used in the following are defined
   in <a href="/doc/html/rfc3550#section-6.3">Section&nbsp;6.3 of [RFC3550]</a>.  The variables T_rr_interval and
   T_rr_last are defined in [<a href="/doc/html/rfc4585" title="&quot;Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)&quot;">RFC4585</a>].

   Each endpoint MUST schedule RTCP transmission independently for each
   of its SSRCs using the regular calculation of tn for the RTP profile
   being used.  Each time the timer tn expires for an SSRC, the endpoint
   MUST perform RTCP timer reconsideration and, if applicable,
   suppression based on T_rr_interval.  If the result indicates that a
   compound RTCP packet is to be sent by that SSRC, and the transmission
   is not an early RTCP packet [<a href="/doc/html/rfc4585" title="&quot;Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)&quot;">RFC4585</a>], then the endpoint SHOULD try
   to aggregate RTCP packets of additional SSRCs that are scheduled in
   the future into the compound RTCP packet before it is sent.  The
   reason to limit or not aggregate due to backwards compatibility
   reasons is discussed in <a href="#section-5.3.1">Section 5.3.1</a>.

   Aggregation proceeds as follows.  The endpoint selects the SSRC that
   has the smallest tn value after the current time, tc, and prepares
   the RTCP packets that SSRC would send if its timer tn expired at tc.
   If those RTCP packets will fit into the compound RTCP packet that is
   being generated, taking into account the path MTU and the previously
   added RTCP packets, then they are added to the compound RTCP packet;
   otherwise, they are discarded.  This process is repeated for each
   SSRC, in order of increasing tn, until the compound RTCP packet is
   full or all SSRCs have been aggregated.  At that point, the compound
   RTCP packet is sent.





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   When the compound RTCP packet is sent, the endpoint MUST update tp,
   tn, and T_rr_last (if applicable) for each SSRC that was included.
   These variables are updated as follows:

   a.  For the first SSRC that reported in the compound RTCP packet, set
       the effective transmission time, tt, of that SSRC to tc.

   b.  For each additional SSRC that reported in the compound RTCP
       packet, calculate the transmission time that SSRC would have had
       if it had not been aggregated into the compound RTCP packet.
       This is derived by taking tn for that SSRC, then performing
       reconsideration and updating tn until tp + T &lt;= tn.  Once this is
       done, set the effective transmission time, tt, for that SSRC to
       the calculated value of tn.  If the RTP/AVPF profile or the RTP/
       SAVPF profile is being used, then suppression based on
       T_rr_interval MUST NOT be used in this calculation.

   c.  Calculate average effective transmission time, tt_avg, for the
       compound RTCP packet based on the tt values for all SSRCs sent in
       the compound RTCP packet.  Set tp for each of the SSRCs sent in
       the compound RTCP packet to tt_avg.  If the RTP/AVPF profile or
       the RTP/SAVPF profile is being used, set T_tt_last for each SSRC
       sent in the compound RTCP packet to tt_avg.

   d.  For each of the SSRCs sent in the compound RTCP packet, calculate
       new tn values based on the updated parameters and the usual RTCP
       timing rules and reschedule the timers.

   When using the RTP/AVPF profile or the RTP/SAVPF profile, the above
   mechanism only attempts to aggregate RTCP packets when the compound
   RTCP packet to be sent is not an early RTCP packet, and hence the
   algorithm in <a href="/doc/html/rfc4585#section-3.5.3">Section&nbsp;3.5.3 of [RFC4585]</a> will control RTCP scheduling.
   If T_rr_interval == 0, or if T_rr_interval != 0 and option 1, 2a, or
   2b of the algorithm are chosen, then the above mechanism updates the
   necessary variables.  However, if the transmission is suppressed per
   option 2c of the algorithm, then tp is updated to tc as aggregation
   has not taken place.

   Reverse reconsideration MUST be performed following <a href="/doc/html/rfc3550#section-6.3.4">Section&nbsp;6.3.4 of
   [RFC3550]</a>.  In some cases, this can lead to the value of tp after
   reverse reconsideration being larger than tc.  This is not a problem,
   and has the desired effect of proportionally pulling the tp value
   towards tc (as well as tn) as the reporting interval shrinks in
   direct proportion the reduced group size.

   The above algorithm has been shown in simulations [<a href="#ref-Sim88" title="&quot;SIMULATION RESULTS FOR MULTI-STREAM&quot;">Sim88</a>] [<a href="#ref-Sim92" title="&quot;Changes in RTP Multi-stream&quot;">Sim92</a>] to
   maintain the inter-RTCP packet transmission time distribution for
   each SSRC and to consume the same amount of bandwidth as



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   non-aggregated RTCP packets.  With this algorithm, the actual
   transmission interval for an SSRC triggering an RTCP compound packet
   transmission is following the regular transmission rules.  The value
   tp is set to somewhere in the interval [0, 1.5/1.21828*Td] ahead of
   tc.  The actual value is the average of one instance of tc and the
   randomized transmission times of the additional SSRCs; thus, the
   lower range of the interval is more probable.  This compensates for
   the bias that is otherwise introduced by picking the shortest tn
   value out of the N SSRCs included in aggregate.

   The algorithm also handles the cases where the number of SSRCs that
   can be included in an aggregated packet varies.  An SSRC that
   previously was aggregated and fails to fit in a packet still has its
   own transmission scheduled according to normal rules.  Thus, it will
   trigger a transmission in due time, or the SSRC will be included in
   another aggregate.  The algorithm&#x27;s behavior under SSRC group size
   changes is as follows:

   RTP sessions where the number of SSRCs is growing:  When the group
      size is growing, Td grows in proportion to the number of new SSRCs
      in the group.  When reconsideration is performed due to expiry of
      the tn timer, that SSRC will reconsider the transmission and with
      a certain probability reschedule the tn timer.  This part of the
      reconsideration algorithm is only impacted by the above algorithm
      having tp values that were in the future instead of set to the
      time of the actual last transmission at the time of updating tp.

   RTP sessions where the number of SSRCs is shrinking:  When the group
      shrinks, reverse reconsideration moves the tp and tn values
      towards tc proportionally to the number of SSRCs that leave the
      session compared to the total number of participants when they
      left.  The setting of the tp value forward in time related to the
      tc could be believed to have negative effect.  However, the reason
      for this setting is to compensate for bias caused by picking the
      shortest tn out of the N aggregated.  This bias remains over a
      reduction in the number of SSRCs.  The reverse reconsideration
      compensates the reduction independently of whether or not
      aggregation is being used.  The negative effect that can occur on
      removing an SSRC is that the most favorable tn belonged to the
      removed SSRC.  The impact of this is limited to delaying the
      transmission, in the worst case, one reporting interval.

   In conclusion, the investigations performed have found no significant
   negative impact on the scheduling algorithm.







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<span class="h3"><a class="selflink" id="section-5.4" href="#section-5.4">5.4</a>.  Use of RTP/AVPF or RTP/SAVPF Feedback</span>

   This section discusses the transmission of RTP/AVPF feedback packets
   when the transmitting endpoint has multiple SSRCs.  The guidelines in
   this section also apply to endpoints using the RTP/SAVPF profile.

<span class="h4"><a class="selflink" id="section-5.4.1" href="#section-5.4.1">5.4.1</a>.  Choice of SSRC for Feedback Packets</span>

   When an RTP/AVPF endpoint has multiple SSRCs, it can choose what SSRC
   to use as the source for the RTCP feedback packets it sends.  Several
   factors can affect that choice:

   o  RTCP feedback packets relating to a particular media type SHOULD
      be sent by an SSRC that receives that media type.  For example,
      when audio and video are multiplexed onto a single RTP session,
      endpoints will use their audio SSRC to send feedback on the audio
      received from other participants.

   o  RTCP feedback packets and RTCP codec control messages that are
      notifications or indications regarding RTP data processed by an
      endpoint MUST be sent from the SSRC used for that RTP data.  This
      includes notifications that relate to a previously received
      request or command [<a href="/doc/html/rfc4585" title="&quot;Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)&quot;">RFC4585</a>][RFC5104].

   o  If separate SSRCs are used to send and receive media, then the
      corresponding SSRC SHOULD be used for feedback, since they have
      differing RTCP bandwidth fractions.  This can also affect the
      consideration of whether or not the SSRC can be used in immediate
      mode.

   o  Some RTCP feedback packet types require consistency in the SSRC
      used.  For example, if a Temporary Maximum Media Stream Bit Rate
      Request (TMMBR) limitation [<a href="/doc/html/rfc5104" title="&quot;Codec Control Messages in the RTP Audio-Visual Profile with Feedback (AVPF)&quot;">RFC5104</a>] is set by an SSRC, the same
      SSRC needs to be used to remove the limitation.

   o  If several SSRCs are suitable for sending feedback, it might be
      desirable to use an SSRC that allows the sending of feedback as an
      early RTCP packet.

   When an RTCP feedback packet is sent as part of a compound RTCP
   packet that aggregates reports from multiple SSRCs, there is no
   requirement that the compound packet contain an SR or RR packet
   generated by the sender of the RTCP feedback packet.  For reduced-
   size RTCP packets, aggregation of RTCP feedback packets from multiple
   sources is not limited further than <a href="/doc/html/rfc5506#section-4.2.2">Section&nbsp;4.2.2 of [RFC5506]</a>.






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<span class="h4"><a class="selflink" id="section-5.4.2" href="#section-5.4.2">5.4.2</a>.  Scheduling an RTCP Feedback Packet</span>

   When an SSRC has a need to transmit a feedback packet in early mode,
   it MUST schedule that packet following the algorithm in <a href="/doc/html/rfc4585#section-3.5">Section&nbsp;3.5
   of [RFC4585]</a> modified as follows:

   o  To determine whether an RTP session is considered to be a point-
      to-point session or a multiparty session, an endpoint MUST count
      the number of distinct RTCP SDES CNAME values used by the SSRCs
      listed in the SSRC field of RTP data packets it receives and in
      the &quot;SSRC of sender&quot; field of RTCP SR, RR, RTPFB, or PSFB packets
      it receives.  An RTP session is considered to be a multiparty
      session if more than one CNAME is used by those SSRCs, unless
      signaling indicates that the session is to be handled as point to
      point or RTCP reporting groups [<a href="#ref-MULTI-STREAM-OPT">MULTI-STREAM-OPT</a>] are used.  If
      RTCP reporting groups are used, an RTP session is considered to be
      a point-to-point session if the endpoint receives only a single
      reporting group and is considered to be a multiparty session if
      multiple reporting groups are received or a combination of
      reporting groups and SSRCs that are not part of a reporting group
      are received.  Endpoints MUST NOT determine whether an RTP session
      is multiparty or point to point based on the type of connection
      (unicast or multicast) used, or on the number of SSRCs received.

   o  When checking if there is already a scheduled compound RTCP packet
      containing feedback messages (Step 2 in <a href="/doc/html/rfc4585#section-3.5.2">Section&nbsp;3.5.2 of
      [RFC4585]</a>), that check MUST be done considering all local SSRCs.

   o  If an SSRC is not allowed to send an early RTCP packet, then the
      feedback message MAY be queued for transmission as part of any
      early or regular scheduled transmission that can occur within the
      maximum useful lifetime of the feedback message (T_max_fb_delay).
      This modifies the behavior in item 4a in <a href="/doc/html/rfc4585#section-3.5.2">Section&nbsp;3.5.2 of
      [RFC4585]</a>.

   The first bullet point above specifies a rule to determine if an RTP
   session is to be considered a point-to-point session or a multiparty
   session.  This rule is straightforward to implement, but is known to
   incorrectly classify some sessions as multiparty sessions.  The known
   problems are as follows:

   Endpoint with multiple synchronization contexts:  An endpoint that is
      part of a point-to-point session can have multiple synchronization
      contexts, for example, due to forwarding an external media source
      into an interactive real-time conversation.  In this case, the
      classification will consider the peer as two endpoints, while the
      actual RTP/RTCP transmission will be under the control of one
      endpoint.



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   Selective Forwarding Middlebox:  The Selective Forwarding Middlebox
      (SFM) as defined in <a href="/doc/html/rfc7667#section-3.7">Section&nbsp;3.7 of [RFC7667]</a> has control over the
      transmission and configurations between itself and each peer
      endpoint individually.  It also fully controls the RTCP packets
      being forwarded between the individual legs.  Thus, this type of
      middlebox can be compared to the RTP mixer, which uses its own
      SSRCs to mix or select the media it forwards, that will be
      classified as a point-to-point RTP session by the above rule.

   In the above cases, it is very reasonable to use RTCP reporting
   groups [<a href="#ref-MULTI-STREAM-OPT">MULTI-STREAM-OPT</a>].  If that extension is used, an endpoint
   can indicate that the multitude of CNAMEs are in fact under a single
   endpoint or middlebox control by using only a single reporting group.

   The above rules will also classify some sessions where the endpoint
   is connected to an RTP mixer as being point to point.  For example,
   the mixer could act as gateway to an RTP session based on Any Source
   Multicast for the discussed endpoint.  However, this will, in most
   cases, be okay, as the RTP mixer provides separation between the two
   parts of the session.  The responsibility falls on the mixer to act
   accordingly in each domain.

   Finally, we note that signaling mechanisms could be defined to
   override the rules when they would result in the wrong
   classification.

<span class="h2"><a class="selflink" id="section-6" href="#section-6">6</a>.  Adding and Removing SSRCs</span>

   The set of SSRCs present in a single RTP session can vary over time
   due to changes in the number of endpoints in the session or due to
   changes in the number or type of RTP streams being sent.

   Every endpoint in an RTP session will have at least one SSRC that it
   uses for RTCP reporting, and for sending media if desired.  It can
   also have additional SSRCs, for sending extra media sources or for
   additional RTCP reporting.  If the set of media sources being sent
   changes, then the set of SSRCs being sent will change.  Changes in
   the media format or clock rate might also require changes in the set
   of SSRCs used.  An endpoint can also have more SSRCs than it has
   active RTP streams, and send RTCP relating to SSRCs that are not
   currently sending RTP data packets so that its peers are aware of the
   SSRCs, and have the associated context (e.g., clock synchronization
   and an SDES CNAME) in place to be able to play out media as soon as
   they becomes active.

   In the following, we describe some considerations around adding and
   removing RTP streams and their associated SSRCs.




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<span class="grey"><a href="/doc/html/rfc8108">RFC 8108</a>        Multiple Media Streams in an RTP Session      March 2017</span>


<span class="h3"><a class="selflink" id="section-6.1" href="#section-6.1">6.1</a>.  Adding RTP Streams</span>

   When an endpoint joins an RTP session, it can have zero, one, or more
   RTP streams it will send, or that it is prepared to send.  If it has
   no RTP stream it plans to send, it still needs an SSRC that will be
   used to send RTCP feedback.  If it will send one or more RTP streams,
   it will need the corresponding number of SSRC values.  The SSRCs used
   by an endpoint are made known to other endpoints in the RTP session
   by sending RTP and RTCP packets.  SSRCs can also be signaled using
   non-RTP means (e.g., [<a href="/doc/html/rfc5576" title="&quot;Source-Specific Media Attributes in the Session Description Protocol (SDP)&quot;">RFC5576</a>]).  Unless restricted by signaling, an
   endpoint can, at any time, send an additional RTP stream, identified
   by a new SSRC (this might be associated with a signaling event, but
   that is outside the scope of this memo).  This makes the new SSRC
   visible to the other endpoints in the session, since they share the
   single SSRC space inherent in the definition of an RTP session.

   An endpoint that has never sent an RTP stream will have an SSRC that
   it uses for RTCP reporting.  If that endpoint wants to start sending
   an RTP stream, it is RECOMMENDED that it use its existing SSRC for
   that stream, since otherwise the participant count in the RTP session
   will be unnecessarily increased, leading to a longer RTCP reporting
   interval and larger RTCP reports due to cross reporting.  If the
   endpoint wants to start sending more than one RTP stream, it will
   need to generate a new SSRC for the second and any subsequent RTP
   streams.

   An endpoint that has previously stopped sending an RTP stream, and
   that wants to start sending a new RTP stream, cannot generally reuse
   the existing SSRC, and often needs to generate a new SSRC, because an
   SSRC cannot change media type (e.g., audio to video) or RTP timestamp
   clock rate [<a href="/doc/html/rfc7160" title="&quot;Support for Multiple Clock Rates in an RTP Session&quot;">RFC7160</a>] and because the SSRC might be associated with a
   particular semantic by the application (note: an RTP stream can pause
   and restart using the same SSRC, provided RTCP is sent for that SSRC
   during the pause; these rules only apply to new RTP streams reusing
   an existing SSRC).

<span class="h3"><a class="selflink" id="section-6.2" href="#section-6.2">6.2</a>.  Removing RTP Streams</span>

   An SSRC is removed from an RTP session in one of two ways.  When an
   endpoint stops sending RTP and RTCP packets using an SSRC, then that
   SSRC will eventually time out as described in <a href="/doc/html/rfc3550#section-6.3.5">Section&nbsp;6.3.5 of
   [RFC3550]</a>.  Alternatively, an SSRC can be explicitly removed from use
   by sending an RTCP BYE packet as described in <a href="/doc/html/rfc3550#section-6.3.7">Section&nbsp;6.3.7 of
   [RFC3550]</a>.  It is RECOMMENDED that SSRCs be removed from use by
   sending an RTCP BYE packet.  Note that [<a href="/doc/html/rfc3550" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">RFC3550</a>] requires that the
   RTCP BYE SHOULD be the last RTP/RTCP packet sent in the RTP session





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   for an SSRC.  If an endpoint needs to restart an RTP stream after
   sending an RTCP BYE for its SSRC, it needs to generate a new SSRC
   value for that stream.

   The finality of sending RTCP BYE means that endpoints need to
   consider if the ceasing of transmission of an RTP stream is temporary
   or permanent.  Temporary suspension of media transmission using a
   particular RTP stream (SSRC) needs to maintain that SSRC as an active
   participant, by continuing RTCP transmission for it.  That way the
   media sending can be resumed immediately, knowing that the context is
   in place.  When permanently halting transmission, a participant needs
   to send an RTCP BYE to allow the other participants to use the RTCP
   bandwidth resources and clean up their state databases.

   An endpoint that ceases transmission of all its RTP streams but
   remains in the RTP session MUST maintain at least one SSRC that is to
   be used for RTCP reporting and feedback (i.e., it cannot send a BYE
   for all SSRCs, but needs to retain at least one active SSRC).  As
   some Feedback packets can be bound to media type, there might be a
   need to maintain one SSRC per media type within an RTP session.  An
   alternative can be to create a new SSRC to use for RTCP reporting and
   feedback.  However, to avoid the perception that an endpoint drops
   completely out of an RTP session, such a new SSRC ought to be
   established first -- before terminating all the existing SSRCs.

<span class="h2"><a class="selflink" id="section-7" href="#section-7">7</a>.  RTCP Considerations for Streams with Disparate Rates</span>

   An RTP session has a single set of parameters that configure the
   session bandwidth.  These are the RTCP sender and receiver fractions
   (e.g., the SDP &quot;b=RR:&quot; and &quot;b=RS:&quot; lines [<a href="/doc/html/rfc3556" title="&quot;Session Description Protocol (SDP) Bandwidth Modifiers for RTP Control Protocol (RTCP) Bandwidth&quot;">RFC3556</a>]) and the
   parameters of the RTP/AVPF profile [<a href="/doc/html/rfc4585" title="&quot;Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)&quot;">RFC4585</a>] (e.g., trr-int) if that
   profile (or its secure extension, RTP/SAVPF [<a href="/doc/html/rfc5124" title="&quot;Extended Secure RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/SAVPF)&quot;">RFC5124</a>]) is used.  As a
   consequence, the base RTCP reporting interval, before randomization,
   will be the same for every sending SSRC in an RTP session.
   Similarly, every receiving SSRC in an RTP session will have the same
   base reporting interval, although this can differ from the reporting
   interval chosen by sending SSRCs.  This uniform RTCP reporting
   interval for all SSRCs can result in RTCP reports being sent more
   often, or too seldom, than is considered desirable for an RTP stream.

   For example, consider a scenario in which an audio flow sending at
   tens of kilobits per second is multiplexed into an RTP session with a
   multi-megabit high-quality video flow.  If the session bandwidth is
   configured based on the video sending rate, and the default RTCP
   bandwidth fraction of 5% of the session bandwidth is used, it is
   likely that the RTCP bandwidth will exceed the audio sending rate.
   If the reduced minimum RTCP interval described in <a href="/doc/html/rfc3550#section-6.2">Section&nbsp;6.2 of
   [RFC3550]</a> is then used in the session, as appropriate for video where



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   rapid feedback on damaged I-frames is wanted, the uniform reporting
   interval for all senders could mean that audio sources are expected
   to send RTCP packets more often than they send audio data packets.
   This bandwidth mismatch can be reduced by careful tuning of the RTCP
   parameters, especially trr_int when the RTP/AVPF profile is used, but
   cannot be avoided entirely as it is inherent in the design of the
   RTCP timing rules, and affects all RTP sessions that contain flows
   with greatly mismatched bandwidth.

   Different media rates or desired RTCP behaviors can also occur with
   SSRCs carrying the same media type.  A common case in multiparty
   conferencing is when a small number of video streams are shown in
   high resolution, while the others are shown as low-resolution
   thumbnails, with the choice of which is shown in high resolution
   being voice-activity controlled.  Here the differences are both in
   actual media rate and in choices for what feedback messages might be
   needed.  Other examples of differences that can exist are due to the
   intended usage of a media source.  A media source carrying the video
   of the speaker in a conference is different from a document camera.
   Basic parameters that can differ in this case are frame-rate,
   acceptable end-to-end delay, and the Signal-to-Noise Ratio (SNR)
   fidelity of the image.  These differences affect not only the needed
   bitrates, but also possible transmission behaviors, usable repair
   mechanisms, what feedback messages the control and repair requires,
   the transmission requirements on those feedback messages, and
   monitoring of the RTP stream delivery.  Other similar scenarios can
   also exist.

   Sending multiple media types in a single RTP session causes that
   session to contain more SSRCs than if each media type was sent in a
   separate RTP session.  For example, if two participants each send an
   audio and a video RTP stream in a single RTP session, that session
   will comprise four SSRCs; but if separate RTP sessions had been used
   for audio and video, each of those two RTP sessions would comprise
   only two SSRCs.  Hence, sending multiple RTP streams in an RTP
   session increases the amount of cross reporting between the SSRCs, as
   each SSRC reports on all other SSRCs in the session.  This increases
   the size of the RTCP reports, causing them to be sent less often than
   would be the case if separate RTP sessions where used for a given
   RTCP bandwidth.

   Finally, when an RTP session contains multiple media types, it is
   important to note that the RTCP reception quality reports, feedback
   messages, and extended report blocks used might not be applicable to
   all media types.  Endpoints will need to consider the media type of
   each SSRC, and only send or process reports and feedback that apply
   to that particular SSRC and its media type.  Signaling solutions




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   might have shortcomings when it comes to indicating that a particular
   set of RTCP reports or feedback messages only apply to a particular
   media type within an RTP session.

   From an RTCP perspective, therefore, it can be seen that there are
   advantages to using separate RTP sessions for each media source,
   rather than sending multiple media sources in a single RTP session.
   However, these are frequently offset by the need to reduce port use,
   to ease NAT/firewall traversal, achieved by combining media sources
   into a single RTP session.  The following sections consider some of
   the issues with using RTCP in sessions with multiple media sources in
   more detail.

<span class="h3"><a class="selflink" id="section-7.1" href="#section-7.1">7.1</a>.  Timing Out SSRCs</span>

   Various issues have been identified with timing out SSRC values when
   sending multiple RTP streams in an RTP session.

<span class="h4"><a class="selflink" id="section-7.1.1" href="#section-7.1.1">7.1.1</a>.  Problems with the RTP/AVPF T_rr_interval Parameter</span>

   The RTP/AVPF profile includes a method to prevent regular RTCP
   reports from being sent too often.  This mechanism is described in
   <a href="/doc/html/rfc4585#section-3.5.3">Section&nbsp;3.5.3 of [RFC4585]</a>; it is controlled by the T_rr_interval
   parameter.  It works as follows.  When a regular RTCP report is sent,
   a new random value, T_rr_current_interval, is generated, drawn evenly
   in the range 0.5 to 1.5 times T_rr_interval.  If a regular RTCP
   packet is to be sent earlier than T_rr_current_interval seconds after
   the previous regular RTCP packet, and there are no feedback messages
   to be sent, then that regular RTCP packet is suppressed and the next
   regular RTCP packet is scheduled.  The T_rr_current_interval is
   recalculated each time a regular RTCP packet is sent.  The benefit of
   suppression is that it avoids wasting bandwidth when there is nothing
   requiring frequent RTCP transmissions, but still allows utilization
   of the configured bandwidth when feedback is needed.

   Unfortunately, this suppression mechanism skews the distribution of
   the RTCP sending intervals compared to the regular RTCP reporting
   intervals.  The standard RTCP timing rules, including reconsideration
   and the compensation factor, result in the intervals between sending
   RTCP packets having a distribution that is skewed towards the upper
   end of the range [0.5/1.21828, 1.5/1.21828]*Td, where Td is the
   deterministic calculated RTCP reporting interval.  With Td = 5 s,
   this distribution covers the range [2.052 s, 6.156 s].  In
   comparison, the RTP/AVPF suppression rules act in an interval that is
   0.5 to 1.5 times T_rr_interval; for T_rr_interval = 5s, this is
   [2.5 s, 7.5 s].





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   The effect of this is that the time between consecutive RTCP packets
   when using T_rr_interval suppression can become large.  The maximum
   time interval between sending one regular RTCP packet and the next,
   when T_rr_interval is being used, occurs when T_rr_current_interval
   takes its maximum value and a regular RTCP packet is suppressed at
   the end of the suppression period, then the next regular RTCP packet
   is scheduled after its largest possible reporting interval.  Taking
   the worst case of the two intervals gives a maximum time between two
   RTCP reports of 1.5*T_rr_interval + 1.5/1.21828*Td.

   This behavior can be surprising when Td and T_rr_interval have the
   same value.  That is, when T_rr_interval is configured to match the
   regular RTCP reporting interval.  In this case, one might expect that
   regular RTCP packets are sent according to their usual schedule, but
   feedback packets can be sent early.  However, the above-mentioned
   issue results in the RTCP packets actually being sent in the range
   [0.5*Td, 2.731*Td] with a highly non-uniform distribution, rather
   than the range [0.41*Td, 1.23*Td].  This is perhaps unexpected, but
   is not a problem in itself.  However, when coupled with packet loss,
   it raises the issue of premature timeout.

<span class="h4"><a class="selflink" id="section-7.1.2" href="#section-7.1.2">7.1.2</a>.  Avoiding Premature Timeout</span>

   In RTP/AVP [<a href="/doc/html/rfc3550" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">RFC3550</a>] the timeout behavior is simple; it is 5 times
   Td, where Td is calculated with a Tmin value of 5 seconds.  In other
   words, if the configured RTCP bandwidth allows for an average RTCP
   reporting interval shorter than 5 seconds, the timeout is 25 seconds
   of no activity from the SSRC (RTP or RTCP); otherwise, the timeout is
   5 average reporting intervals.

   RTP/AVPF [<a href="/doc/html/rfc4585" title="&quot;Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)&quot;">RFC4585</a>] introduces different timeout behaviors depending
   on the value of T_rr_interval.  When T_rr_interval is 0, it uses the
   same timeout calculation as RTP/AVP.  However, when T_rr_interval is
   non-zero, it replaces Tmin in the timeout calculation, most likely to
   speed up detection of timed out SSRCs.  However, using a non-zero
   T_rr_interval has two consequences for RTP behavior.

   First, due to suppression, the number of RTP and RTCP packets sent by
   an SSRC that is not an active RTP sender can become very low, because
   of the issue discussed in <a href="#section-7.1.1">Section 7.1.1</a>.  As the RTCP packet interval
   can be as long as 2.73*Td, during a 5*Td time period, an endpoint
   might in fact transmit only a single RTCP packet.  The long intervals
   result in fewer RTCP packets, to a point where a single RTCP packet
   loss can sometimes result in timing out an SSRC.

   Second, the RTP/AVPF changes to the timeout rules reduce robustness
   to misconfiguration.  It is common to use RTP/AVPF configured such
   that RTCP packets can be sent frequently to allow rapid feedback;



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   however, this makes timeouts very sensitive to T_rr_interval.  For
   example, if two SSRCs are configured, one with T_rr_interval = 0.1 s
   and the other with T_rr_interval = 0.6 s, then this small difference
   will result in the SSRC with the shorter T_rr_interval timing out the
   other if it stops sending RTP packets, since the other RTCP reporting
   interval is more than five times its own.  When RTP/AVP is used, or
   RTP/AVPF with T_rr_interval = 0, this is a non-issue, as the timeout
   period will be 25 s, and differences between configured RTCP
   bandwidth can only cause premature timeouts when the reporting
   intervals are greater than 5 s and differ by a factor of five.  To
   limit the scope for such problematic misconfiguration, we define an
   update to the RTP/AVPF timeout rules in <a href="#section-7.1.4">Section 7.1.4</a>.

<span class="h4"><a class="selflink" id="section-7.1.3" href="#section-7.1.3">7.1.3</a>.  Interoperability between RTP/AVP and RTP/AVPF</span>

   If endpoints implementing the RTP/AVP and RTP/AVPF profiles (or their
   secure variants) are combined within a single RTP session, and the
   RTP/AVPF endpoints use a non-zero T_rr_interval that is significantly
   below 5 seconds, there is a risk that the RTP/AVPF endpoints will
   prematurely time out the SSRCs of the RTP/AVP endpoints, due to their
   different RTCP timeout rules.  Conversely, if the RTP/AVPF endpoints
   use a T_rr_interval that is significantly larger than 5 seconds,
   there is a risk that the RTP/AVP endpoints will time out the SSRCs of
   the RTP/AVPF endpoints.

   Mixing endpoints using two different RTP profiles within a single RTP
   session is NOT RECOMMENDED.  However, if mixed RTP profiles are used,
   and the RTP/AVPF endpoints are not updated to follow <a href="#section-7.1.4">Section 7.1.4</a> of
   this memo, then the RTP/AVPF session SHOULD be configured to use
   T_rr_interval = 4 seconds to avoid premature timeouts.

   The choice of T_rr_interval = 4 seconds for interoperability might
   appear strange.  Intuitively, this value ought to be 5 seconds, to
   make both the RTP/AVP and RTP/AVPF use the same timeout period.
   However, the behavior outlined in <a href="#section-7.1.1">Section 7.1.1</a> shows that actual
   RTP/AVPF reporting intervals can be longer than expected.  Setting
   T_rr_interval = 4 seconds gives actual RTCP intervals near to those
   expected by RTP/AVP, ensuring interoperability.













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<span class="h4"><a class="selflink" id="section-7.1.4" href="#section-7.1.4">7.1.4</a>.  Updated SSRC Timeout Rules</span>

   To ensure interoperability and avoid premature timeouts, all SSRCs in
   an RTP session MUST use the same timeout behavior.  However, previous
   specifications are inconsistent in this regard.  To avoid
   interoperability issues, this memo updates the timeout rules as
   follows:

   o  For the RTP/AVP, RTP/SAVP, RTP/AVPF, and RTP/SAVPF profiles, the
      timeout interval SHALL be calculated using a multiplier of five
      times the deterministic RTCP reporting interval.  That is, the
      timeout interval SHALL be 5*Td.

   o  For the RTP/AVP, RTP/SAVP, RTP/AVPF, and RTP/SAVPF profiles,
      calculation of Td, for the purpose of calculating the participant
      timeout only, SHALL be done using a Tmin value of 5 seconds and
      not the reduced minimal interval, even if the reduced minimum
      interval is used to calculate RTCP packet transmission intervals.

   This changes the behavior for the RTP/AVPF or RTP/SAVPF profiles when
   T_rr_interval != 0.  Specifically, the first paragraph of
   <a href="/doc/html/rfc4585#section-3.5.4">Section&nbsp;3.5.4 of [RFC4585]</a> is updated to use Tmin instead of
   T_rr_interval in the timeout calculation for RTP/AVPF entities.

<span class="h3"><a class="selflink" id="section-7.2" href="#section-7.2">7.2</a>.  Tuning RTCP Transmissions</span>

   This subsection discusses what tuning can be done to reduce the
   downsides of the shared RTCP packet intervals.  First, what
   possibilities exist for the RTP/AVP [<a href="/doc/html/rfc3551" title="&quot;RTP Profile for Audio and Video Conferences with Minimal Control&quot;">RFC3551</a>] profile are listed
   followed by what additional tools are provided by RTP/AVPF [<a href="/doc/html/rfc4585" title="&quot;Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)&quot;">RFC4585</a>].

<span class="h4"><a class="selflink" id="section-7.2.1" href="#section-7.2.1">7.2.1</a>.  RTP/AVP and RTP/SAVP</span>

   When using the RTP/AVP or RTP/SAVP profiles, the options for tuning
   the RTCP reporting intervals are limited to the RTCP sender and
   receiver bandwidth, and whether the minimum RTCP interval is scaled
   according to the bandwidth.  As the scheduling algorithm includes
   both randomization and reconsideration, one cannot simply calculate
   the expected average transmission interval using the formula for Td
   given in <a href="/doc/html/rfc3550#section-6.3.1">Section&nbsp;6.3.1 of [RFC3550]</a>.  However, by considering the
   inputs to that expression, and the randomization and reconsideration
   rules, we can begin to understand the behavior of the RTCP
   transmission interval.








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   Let&#x27;s start with some basic observations:

   a.  Unless the scaled minimum RTCP interval is used, Td prior to
       randomization and reconsideration can never be less than Tmin.
       The default value of Tmin is 5 seconds.

   b.  If the scaled minimum RTCP interval is used, Td can become as low
       as 360 divided by RTP Session bandwidth in kilobits per second.
       In SDP, the RTP session bandwidth is signaled using a &quot;b=AS&quot;
       line.  An RTP Session bandwidth of 72 kbps results in Tmin being
       5 seconds.  An RTP session bandwidth of 360 kbps of course gives
       a Tmin of 1 second, and to achieve a Tmin equal to once every
       frame for a 25 frame-per-second video stream requires an RTP
       session bandwidth of 9 Mbps.  Use of the RTP/AVPF or RTP/SAVPF
       profile allows more frequent RTCP reports for the same bandwidth,
       as discussed below.

   c.  The value of Td scales with the number of SSRCs and the average
       size of the RTCP reports to keep the overall RTCP bandwidth
       constant.

   d.  The actual transmission interval for a Td value is in the range
       [0.5*Td/1.21828, 1.5*Td/1.21828], and the distribution is skewed,
       due to reconsideration, with the majority of the probability mass
       being above Td.  This means, for example, that for Td = 5 s, the
       actual transmission interval will be distributed in the range
       [2.052 s, 6.156 s], and tending towards the upper half of the
       interval.  Note that Tmin parameter limits the value of Td before
       randomization and reconsideration are applied, so the actual
       transmission interval will cover a range extending below Tmin.

   Given the above, we can calculate the number of SSRCs, n, that an RTP
   session with 5% of the session bandwidth assigned to RTCP can support
   while maintaining Td equal to Tmin.  This will tell us how many RTP
   streams we can report on, keeping the RTCP overhead within acceptable
   bounds.  We make two assumptions that simplify the calculation: that
   all SSRCs are senders, and that they all send compound RTCP packets
   comprising an SR packet with n-1 report blocks, followed by an SDES
   packet containing a 16 octet CNAME value [<a href="/doc/html/rfc7022" title="&quot;Guidelines for Choosing RTP Control Protocol (RTCP) Canonical Names (CNAMEs)&quot;">RFC7022</a>] (such RTCP packets
   will vary in size between 54 and 798 octets depending on n, up to the
   maximum of 31 report blocks that can be included in an SR packet).
   If we put this packet size, and a 5% RTCP bandwidth fraction into the
   RTCP interval calculation in <a href="/doc/html/rfc3550#section-6.3.1">Section&nbsp;6.3.1 of [RFC3550]</a>, and
   calculate the value of n needed to give Td = Tmin for the scaled
   minimum interval, we find n=9 SSRCs can be supported (irrespective of
   the interval, due to the way the reporting interval scales with the
   session bandwidth).  We see that to support more SSRCs without
   changing the scaled minimum interval, we need to increase the RTCP



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   bandwidth fraction from 5%; changing the session bandwidth to a
   higher value would reduce the Tmin.  However, if using the default 5%
   allocation of RTCP bandwidth, an increase will result in more SSRCs
   being supported given a fixed Td target.

   Based on the above, when using the RTP/AVP profile or the RTP/SAVP
   profile, the key limitation for rapid RTCP reporting in small unicast
   sessions is going to be the Tmin value.  The RTP session bandwidth
   configured in RTCP has to be sufficiently high to reach the reporting
   goals the application has following the rules for the scaled minimal
   RTCP interval.

<span class="h4"><a class="selflink" id="section-7.2.2" href="#section-7.2.2">7.2.2</a>.  RTP/AVPF and RTP/SAVPF</span>

   When using RTP/AVPF or RTP/SAVPF, we have a powerful additional tool
   for tuning RTCP transmissions: the T_rr_interval parameter.  Use of
   this parameter allows short RTCP reporting intervals; alternatively
   it gives the ability to sent frequent RTCP feedback without sending
   frequent regular RTCP reports.

   The use of the RTP/AVPF or RTP/SAVPF profile with T_rr_interval set
   to a value greater than zero but smaller than Tmin allows more
   frequent RTCP feedback than the RTP/AVP or RTP/SAVP profiles, for a
   given RTCP bandwidth.  This happens because Tmin is set to zero after
   the transmission of the initial RTCP report, causing the reporting
   interval for later packet to be determined by the usual RTCP
   bandwidth-based calculation, with Tmin=0, and the T_rr_interval.
   This has the effect that we are no longer restricted by the minimal
   interval (whether the default 5-second minimum or the reduced minimum
   interval).  Rather, the RTCP bandwidth and the T_rr_interval are the
   governing factors, allowing faster feedback.  Applications that care
   about rapid regular RTCP feedback ought to consider using the RTP/
   AVPF or RTP/SAVPF profile, even if they don&#x27;t use the feedback
   features of that profile.

   The use of the RTP/AVPF or RTP/SAVPF profile allows RTCP feedback
   packets to be sent frequently, without also requiring regular RTCP
   reports to be sent frequently, since T_rr_interval limits the rate at
   which regular RTCP packets can be sent, while still permitting RTCP
   feedback packets to be sent.  Applications that can use feedback
   packets for some RTP streams, e.g., video streams, but don&#x27;t want
   frequent regular reporting for other RTP streams, can configure the
   T_rr_interval to a value so that the regular reporting for both audio
   and video is at a level that is considered acceptable for the audio.
   They could then use feedback packets, which will include RTCP SR/RR
   packets unless reduced size RTCP feedback packets [<a href="/doc/html/rfc5506" title="&quot;Support for Reduced-Size Real-Time Transport Control Protocol (RTCP): Opportunities and Consequences&quot;">RFC5506</a>] are used,





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<span class="grey"><a href="/doc/html/rfc8108">RFC 8108</a>        Multiple Media Streams in an RTP Session      March 2017</span>


   for the video reporting.  This allows the available RTCP bandwidth to
   be devoted on the feedback that provides the most utility for the
   application.

   Using T_rr_interval still requires one to determine suitable values
   for the RTCP bandwidth value.  Indeed, it might make this choice even
   more important, as this is more likely to affect the RTCP behavior
   and performance than when using the RTP/AVP or RTP/SAVP profile, as
   there are fewer limitations affecting the RTCP transmission.

   When T_rr_interval is non-zero, there are configurations that need to
   be avoided.  If the RTCP bandwidth chosen is such that the Td value
   is smaller than, but close to, T_rr_interval, then the actual regular
   RTCP packet transmission interval can become very large, as discussed
   in <a href="#section-7.1.1">Section 7.1.1</a>.  Therefore, for configuration where one intends to
   have Td smaller than T_rr_interval, then Td is RECOMMENDED to be
   targeted at values less than 1/4th of T_rr_interval, which results in
   the range becoming [0.5*T_rr_interval, 1.81*T_rr_interval].

   With the RTP/AVPF or RTP/SAVPF profiles, using T_rr_interval = 0 has
   utility and results in a behavior where the RTCP transmission is only
   limited by the bandwidth, i.e., no Tmin limitations at all.  This
   allows more frequent regular RTCP reporting than can be achieved
   using the RTP/AVP profile.  Many configurations of RTCP will not
   consume all the bandwidth that they have been configured to use, but
   this configuration will consume what it has been given.  Note that
   the same behavior will be achieved as long as T_rr_interval is
   smaller than 1/3 of Td as that prevents T_rr_interval from affecting
   the transmission.

   There exists no method for using different regular RTCP reporting
   intervals depending on the media type or individual RTP stream, other
   than using a separate RTP session for each type or stream.

<span class="h2"><a class="selflink" id="section-8" href="#section-8">8</a>.  Security Considerations</span>

   When using the secure RTP protocol (RTP/SAVP) [<a href="/doc/html/rfc3711" title="&quot;The Secure Real-time Transport Protocol (SRTP)&quot;">RFC3711</a>], or the
   secure variant of the feedback profile (RTP/SAVPF) [<a href="/doc/html/rfc5124" title="&quot;Extended Secure RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/SAVPF)&quot;">RFC5124</a>], the
   cryptographic context of a compound secure RTCP packet is the SSRC of
   the sender of the first RTCP (sub-)packet.  This could matter in some
   cases, especially for keying mechanisms such as MIKEY [<a href="/doc/html/rfc3830" title="&quot;MIKEY: Multimedia Internet KEYing&quot;">RFC3830</a>] that
   allow use of per-SSRC keying.

   Otherwise, the standard security considerations of RTP apply; sending
   multiple RTP streams from a single endpoint in a single RTP session
   does not appear to have different security consequences than sending
   the same number of RTP streams spread across different RTP sessions.




<span class="grey">Lennox, et al.               Standards Track                   [Page 25]</span></pre>
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<span class="h2"><a class="selflink" id="section-9" href="#section-9">9</a>.  References</span>

<span class="h3"><a class="selflink" id="section-9.1" href="#section-9.1">9.1</a>.  Normative References</span>

   [<a id="ref-RFC2119">RFC2119</a>]  Bradner, S., &quot;Key words for use in RFCs to Indicate
              Requirement Levels&quot;, <a href="/doc/html/bcp14">BCP 14</a>, <a href="/doc/html/rfc2119">RFC 2119</a>,
              DOI 10.17487/RFC2119, March 1997,
              &lt;<a href="https://www.rfc-editor.org/info/rfc2119">http://www.rfc-editor.org/info/rfc2119</a>&gt;.

   [<a id="ref-RFC3550">RFC3550</a>]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, &quot;RTP: A Transport Protocol for Real-Time
              Applications&quot;, STD 64, <a href="/doc/html/rfc3550">RFC 3550</a>, DOI 10.17487/RFC3550,
              July 2003, &lt;<a href="https://www.rfc-editor.org/info/rfc3550">http://www.rfc-editor.org/info/rfc3550</a>&gt;.

   [<a id="ref-RFC3711">RFC3711</a>]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
              Norrman, &quot;The Secure Real-time Transport Protocol (SRTP)&quot;,
              <a href="/doc/html/rfc3711">RFC 3711</a>, DOI 10.17487/RFC3711, March 2004,
              &lt;<a href="https://www.rfc-editor.org/info/rfc3711">http://www.rfc-editor.org/info/rfc3711</a>&gt;.

   [<a id="ref-RFC4585">RFC4585</a>]  Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
              &quot;Extended RTP Profile for Real-time Transport Control
              Protocol (RTCP)-Based Feedback (RTP/AVPF)&quot;, <a href="/doc/html/rfc4585">RFC 4585</a>,
              DOI 10.17487/RFC4585, July 2006,
              &lt;<a href="https://www.rfc-editor.org/info/rfc4585">http://www.rfc-editor.org/info/rfc4585</a>&gt;.

   [<a id="ref-RFC5124">RFC5124</a>]  Ott, J. and E. Carrara, &quot;Extended Secure RTP Profile for
              Real-time Transport Control Protocol (RTCP)-Based Feedback
              (RTP/SAVPF)&quot;, <a href="/doc/html/rfc5124">RFC 5124</a>, DOI 10.17487/RFC5124, February
              2008, &lt;<a href="https://www.rfc-editor.org/info/rfc5124">http://www.rfc-editor.org/info/rfc5124</a>&gt;.

   [<a id="ref-RFC5506">RFC5506</a>]  Johansson, I. and M. Westerlund, &quot;Support for Reduced-Size
              Real-Time Transport Control Protocol (RTCP): Opportunities
              and Consequences&quot;, <a href="/doc/html/rfc5506">RFC 5506</a>, DOI 10.17487/RFC5506, April
              2009, &lt;<a href="https://www.rfc-editor.org/info/rfc5506">http://www.rfc-editor.org/info/rfc5506</a>&gt;.

<span class="h3"><a class="selflink" id="section-9.2" href="#section-9.2">9.2</a>.  Informative References</span>

   [<a id="ref-CLUE-FRAME">CLUE-FRAME</a>]
              Duckworth, M., Ed., Pepperell, A., and S. Wenger,
              &quot;Framework for Telepresence Multi-Streams&quot;, Work in
              Progress, <a href="/doc/html/draft-ietf-clue-framework-25">draft-ietf-clue-framework-25</a>, January 2016.

   [<a id="ref-MULTI-RTP">MULTI-RTP</a>]
              Westerlund, M., Perkins, C., and J. Lennox, &quot;Sending
              Multiple Types of Media in a Single RTP Session&quot;, Work in
              Progress, <a href="/doc/html/draft-ietf-avtcore-multi-media-rtp-session-13">draft-ietf-avtcore-multi-media-rtp-session-13</a>,
              December 2015.




<span class="grey">Lennox, et al.               Standards Track                   [Page 26]</span></pre>
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   [<a id="ref-MULTI-STREAM-OPT">MULTI-STREAM-OPT</a>]
              Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,
              &quot;Sending Multiple Media Streams in a Single RTP Session:
              Grouping RTCP Reception Statistics and Other Feedback&quot;,
              Work in Progress, <a href="/doc/html/draft-ietf-avtcore-rtp-multi-stream-optimisation-12">draft-ietf-avtcore-rtp-multi-</a>
              <a href="/doc/html/draft-ietf-avtcore-rtp-multi-stream-optimisation-12">stream-optimisation-12</a>, March 2016.

   [<a id="ref-RFC3390">RFC3390</a>]  Allman, M., Floyd, S., and C. Partridge, &quot;Increasing TCP&#x27;s
              Initial Window&quot;, <a href="/doc/html/rfc3390">RFC 3390</a>, DOI 10.17487/RFC3390, October
              2002, &lt;<a href="https://www.rfc-editor.org/info/rfc3390">http://www.rfc-editor.org/info/rfc3390</a>&gt;.

   [<a id="ref-RFC3551">RFC3551</a>]  Schulzrinne, H. and S. Casner, &quot;RTP Profile for Audio and
              Video Conferences with Minimal Control&quot;, STD 65, <a href="/doc/html/rfc3551">RFC 3551</a>,
              DOI 10.17487/RFC3551, July 2003,
              &lt;<a href="https://www.rfc-editor.org/info/rfc3551">http://www.rfc-editor.org/info/rfc3551</a>&gt;.

   [<a id="ref-RFC3556">RFC3556</a>]  Casner, S., &quot;Session Description Protocol (SDP) Bandwidth
              Modifiers for RTP Control Protocol (RTCP) Bandwidth&quot;,
              <a href="/doc/html/rfc3556">RFC 3556</a>, DOI 10.17487/RFC3556, July 2003,
              &lt;<a href="https://www.rfc-editor.org/info/rfc3556">http://www.rfc-editor.org/info/rfc3556</a>&gt;.

   [<a id="ref-RFC3830">RFC3830</a>]  Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K.
              Norrman, &quot;MIKEY: Multimedia Internet KEYing&quot;, <a href="/doc/html/rfc3830">RFC 3830</a>,
              DOI 10.17487/RFC3830, August 2004,
              &lt;<a href="https://www.rfc-editor.org/info/rfc3830">http://www.rfc-editor.org/info/rfc3830</a>&gt;.

   [<a id="ref-RFC4588">RFC4588</a>]  Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
              Hakenberg, &quot;RTP Retransmission Payload Format&quot;, <a href="/doc/html/rfc4588">RFC 4588</a>,
              DOI 10.17487/RFC4588, July 2006,
              &lt;<a href="https://www.rfc-editor.org/info/rfc4588">http://www.rfc-editor.org/info/rfc4588</a>&gt;.

   [<a id="ref-RFC5104">RFC5104</a>]  Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
              &quot;Codec Control Messages in the RTP Audio-Visual Profile
              with Feedback (AVPF)&quot;, <a href="/doc/html/rfc5104">RFC 5104</a>, DOI 10.17487/RFC5104,
              February 2008, &lt;<a href="https://www.rfc-editor.org/info/rfc5104">http://www.rfc-editor.org/info/rfc5104</a>&gt;.

   [<a id="ref-RFC5576">RFC5576</a>]  Lennox, J., Ott, J., and T. Schierl, &quot;Source-Specific
              Media Attributes in the Session Description Protocol
              (SDP)&quot;, <a href="/doc/html/rfc5576">RFC 5576</a>, DOI 10.17487/RFC5576, June 2009,
              &lt;<a href="https://www.rfc-editor.org/info/rfc5576">http://www.rfc-editor.org/info/rfc5576</a>&gt;.

   [<a id="ref-RFC6190">RFC6190</a>]  Wenger, S., Wang, Y., Schierl, T., and A. Eleftheriadis,
              &quot;RTP Payload Format for Scalable Video Coding&quot;, <a href="/doc/html/rfc6190">RFC 6190</a>,
              DOI 10.17487/RFC6190, May 2011,
              &lt;<a href="https://www.rfc-editor.org/info/rfc6190">http://www.rfc-editor.org/info/rfc6190</a>&gt;.






<span class="grey">Lennox, et al.               Standards Track                   [Page 27]</span></pre>
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   [<a id="ref-RFC6928">RFC6928</a>]  Chu, J., Dukkipati, N., Cheng, Y., and M. Mathis,
              &quot;Increasing TCP&#x27;s Initial Window&quot;, <a href="/doc/html/rfc6928">RFC 6928</a>,
              DOI 10.17487/RFC6928, April 2013,
              &lt;<a href="https://www.rfc-editor.org/info/rfc6928">http://www.rfc-editor.org/info/rfc6928</a>&gt;.

   [<a id="ref-RFC7022">RFC7022</a>]  Begen, A., Perkins, C., Wing, D., and E. Rescorla,
              &quot;Guidelines for Choosing RTP Control Protocol (RTCP)
              Canonical Names (CNAMEs)&quot;, <a href="/doc/html/rfc7022">RFC 7022</a>, DOI 10.17487/RFC7022,
              September 2013, &lt;<a href="https://www.rfc-editor.org/info/rfc7022">http://www.rfc-editor.org/info/rfc7022</a>&gt;.

   [<a id="ref-RFC7160">RFC7160</a>]  Petit-Huguenin, M. and G. Zorn, Ed., &quot;Support for Multiple
              Clock Rates in an RTP Session&quot;, <a href="/doc/html/rfc7160">RFC 7160</a>,
              DOI 10.17487/RFC7160, April 2014,
              &lt;<a href="https://www.rfc-editor.org/info/rfc7160">http://www.rfc-editor.org/info/rfc7160</a>&gt;.

   [<a id="ref-RFC7667">RFC7667</a>]  Westerlund, M. and S. Wenger, &quot;RTP Topologies&quot;, <a href="/doc/html/rfc7667">RFC 7667</a>,
              DOI 10.17487/RFC7667, November 2015,
              &lt;<a href="https://www.rfc-editor.org/info/rfc7667">http://www.rfc-editor.org/info/rfc7667</a>&gt;.

   [<a id="ref-SDP-BUNDLE">SDP-BUNDLE</a>]
              Holmberg, C., Alvestrand, H., and C. Jennings,
              &quot;Negotiating Media Multiplexing Using the Session
              Description Protocol (SDP)&quot;, Work in Progress,
              <a href="/doc/html/draft-ietf-mmusic-sdp-bundle-negotiation-36">draft-ietf-mmusic-sdp-bundle-negotiation-36</a>, October 2016.

   [<a id="ref-Sim88">Sim88</a>]    Westerlund, M., &quot;SIMULATION RESULTS FOR MULTI-STREAM&quot;,
              IETF 88 Proceedings, November 2013,
              &lt;<a href="https://www.ietf.org/proceedings/88/slides/slides-88-avtcore-0.pdf">https://www.ietf.org/proceedings/88/slides/</a>
              <a href="https://www.ietf.org/proceedings/88/slides/slides-88-avtcore-0.pdf">slides-88-avtcore-0.pdf</a>&gt;.

   [<a id="ref-Sim92">Sim92</a>]    Westerlund, M., Lennox, J., Perkins, C., and Q. Wu,
              &quot;Changes in RTP Multi-stream&quot;, IETF 92 Proceedings, March
              2015, &lt;<a href="https://www.ietf.org/proceedings/92/slides/slides-92-avtcore-0.pdf">https://www.ietf.org/proceedings/92/slides/</a>
              <a href="https://www.ietf.org/proceedings/92/slides/slides-92-avtcore-0.pdf">slides-92-avtcore-0.pdf</a>&gt;.

















<span class="grey">Lennox, et al.               Standards Track                   [Page 28]</span></pre>
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<span class="grey"><a href="/doc/html/rfc8108">RFC 8108</a>        Multiple Media Streams in an RTP Session      March 2017</span>


Acknowledgments

   The authors like to thank Harald Alvestrand and everyone else who has
   been involved in the development of this document.

Authors&#x27; Addresses

   Jonathan Lennox
   Vidyo, Inc.
   433 Hackensack Avenue
   Seventh Floor
   Hackensack, NJ  07601
   United States of America

   Email: jonathan@vidyo.com


   Magnus Westerlund
   Ericsson
   Farogatan 2
   SE-164 80 Kista
   Sweden

   Phone: +46 10 714 82 87
   Email: magnus.westerlund@ericsson.com


   Qin Wu
   Huawei
   101 Software Avenue, Yuhua District
   Nanjing, Jiangsu 210012
   China

   Email: bill.wu@huawei.com


   Colin Perkins
   University of Glasgow
   School of Computing Science
   Glasgow  G12 8QQ
   United Kingdom

   Email: csp@csperkins.org








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